similar to: Re: Contents of Asterisk-Users digest...

Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software.

Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works

Dear Asterisk Fans, I'm an Asterisk consultant in Argentina and want to make an spanish wiki (something like voip-info.org). I have the idea and some concepts about this project. It won't be a comercial project, it would be free and it's target would be spanish talking asterisk enthusiasts. I'm wrinting these for the sake of finding contributors, people who want to help me

Hello everybody, I'm installing an Asterisk 1.2.7.1 with GNUDialer 0.98-puff18. It also has zaptel from CVS. My FXO is an X100P Clone. The agents from GNUDialer log ok, and everything is fine until the GNUDialer makes a call, as soon as it engages (the phone starts to ring) asterisk crashes with these messages: > Channel Zap/1-1 was answered. -- Executing

I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Open your mind, use open source.

Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal.

Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of

Hello everyone! In my humble try of creating a Blog, I've made this: http://fameal.blogdns.org. By now, it's hosted in my own server but shortly it'll be moved to a serious hosting. All post are written in spanish, so it's only for spanish talking people, I will try to make it grow and have english articles. If someone is interested in writing in english (obiously I would help) I

It's bad enough that people insert long config/log files/dumps into messages to this list, though it's convenient. But when people include the entire redundant content in a quote in reply, it's really a waste. The digest messages are hard enough to navigate, even with an intro index, before making 95% of their content a redundant quoted dump. I know we've all got lots of bandwidth

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP

Well, asking to dev-icecast list, James Scoles told me that icecast by itself can't do what i need. So he recomended me to use "liquidsoap" to do the rebroadcast. It is working just fine, because i didn't have time to discover some new thing yet. I wish to share with all of you. Thanks. 2011/4/21 Sascha Bieler <sascha.bieler at radiogong.de>: > As far as I know you

28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the

Yes, you have to setup a icecast relay server. That's all. -----Original Message----- From: icecast-bounces at xiph.org [mailto:icecast-bounces at xiph.org] On Behalf Of Facundo Su?rez Sent: Thursday, April 21, 2011 6:44 AM To: icecast at xiph.org Subject: [Icecast] repeater Hi all there. I am trying to set up an icecast2 server (wich is already running) to send some stream mount point to

hi facundo Am Thu, 21 Apr 2011 11:24:14 -0300 schrieb Facundo Su?rez <suarez.jf at gmail.com>: > Hi all there. > > I am trying to set up an icecast2 server (wich is already running) to > send some stream mount point to another server. I mean, my server is > streaming at /radio1.mp3 mount point. We can listen it at > http://myserver:8000/radio1.mp3; so, i want to know if

It looks like you have the first extry of the [incoming] context in extensions.conf commented out Hi all, I am resending this message, so far no one has helped me with this incoming call issue. there is no problem with outbound call but there is no inbound call to my sip phone. the only message I get when I call from pstn is "unable to create local channel for call forward to

Dear list: I have the typical one way audio problem, as far as i know it's a nating problem, my hosts inside my lan can call to outside internet hosts, but can't listen a thing, i read a lot about sip and rtp and protocols and the problem it seems to be with NAT, this is the config i put on my sip.conf file about nat: externhost=sip.server.com.ar > my server name on the

Just checked my figures, and I mean 0.5%-0.7%. Anyway, it is the resulting clicks that are the problem. Any help still appreciated. David -----Original Message----- From: David Brazier Sent: 13 March 2007 00:33 To: asterisk-users@lists.digium.com Subject: Playback 5% Too Fast? Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI

David Brazier wrote: > Hi > > The latest SVN trunk for speex has changed the SpeexPreprocessState to > an opaque structure, for jolly good software engineering reasons. > However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) > relies on some members of this. It uses speech_prob to detect when > there is enough speech to consider AAGC and then loudness2 to

Hi: Is there an easy way to add a HP 5si printer using a Jetdirect interface, to share with my lan?? Maybe some config help?? Thanks a lot -- Facundo Agustin Barrera IT Management. Buenos Aires - Argentina.

Hi list: First post, hope find a solution.... This is my problem, i've got some chm (Microsoft html help files) and i can't see them from my host clients (windows) i can open them but i cant see them... this is the error log from samba: facu (192.168.0.48) couldn't find service e-books - oreilly &amp; cisco smbd/service.c:make_connection(798) Any ideas? Many thanks.